This issue generally only applies if a SIP call arrives on the PABX, and is routed back out to the same SIP provider without stopping on the PABX.
The issue is that SIP requires one party or the other to provide audio to act as a clocking source, but some SIP endpoints do not provide the initial audio under any circumstances, so the audio can never get started.
- This can be rectified by adding a short burst of “silent audio” at the start of the call. Most simply this can be done by setting a “Welcome message” on the extension that handles the call. A sample of audio is provided at the link below that can be uploaded to a spare IVR audio slot and used for this purpose.
20ms.alaw silence file (Right-click and save-as)
- Alternatively, a MUCH bettersolution is to ensure that all of the UDP port numbers listed on the firewall ports FAQ have been opened from all SIP providers. The most common cause of this issue is that the firewall is allowing stateful-only access to SIP, and not permanent access.